[asterisk-users] Unable to find a codec translation path from ulaw|h261 to slin

[Mike Diehl]

I'm doing some final check-outs before upgrading from 1.4.x to 1.6.x and I've 
encountered a problem playing back a .wav file to an Ekiga client:

My dialplan looks like:

exten => 730,1,answer
exten => 730,n,playback(/home/phones/common/moh/moha/Sovereign)
exten => 730,n,hangup

Sovereign.wav is a .wav file that plays nicely on my 1.4 server.

Here is what the console displays:

-- Executing [730 at customers:2] Playback("SIP/user_xxx-00000012", 
"/home/phones/common/moh/moha/Sovereign") in new stack
Unable to find a codec translation path from 0x40004 (ulaw|h261) to 0x40 
(slin)
Unable to open /home/phones/common/moh/moha/Sovereign (format 0x40004 (ulaw|
h261)): No such file or directory ast_streamfile failed on 
SIP/user_xxx-00000012 for /home/phones/common/moh/moha/Sovereign

I was under the impression that I didn't have to do anything to get slin 
support.

when I do: module show like codec_
I get:

Module                         Description                              Use                                                                                
codec_a_mu.so                  A-law and Mulaw direct Coder/Decoder     0                                                                                       
codec_gsm.so                   GSM Coder/Decoder                        0
codec_ulaw.so                  mu-Law Coder/Decoder                     0                                                                                       

I'm assuming use=0 because the server is idle.

I've got allow = all in my sip.conf file.

Anyway, does anyone have an idea on how to resolve this?

-- 

Take care and have fun,
Mike Diehl.